No incoming audio on outbound calls

General Help
1

Hello

We’ve facing here a very strange situation with sipgate in germany.
When we make outbound calls, we have no incoming audio, but ougoing.

When someone calls us from outside (inbound call) audio works in both directions.
Event the support of sigate couldn’t help us.
It seems not to be a firewall or nat related issue because, we have an other trunk running on the same freepbx without any problems.

Tried all possible diffrent trunk configuration without any success yet.

Has someone an idea?

Regards

2

Check your router and make sure you can turn off and have turned SIP-ALG in your border router.

3

There is no sip alg at all on the router/firewall. Its direct 1:1 nat from a specific ip to the FreePBX Server and the regarding rules for sip/5160/5061 and rtp.

4

Can you post here the call flow pcap?

For both incoming and outgoing call (for the working and non working scenarion - infront of the same SIP provider)

John

5

Are you using the same trunk for inbound and outbound calls? If not maybe look at the outbound trunk settings. Try changing the codec to g711 and add nat=yes, keepalive=yes

6

Good point - I have the same problem when I try to call my own PBX from my PBX (dial the office line from my desk phone) and I don’t think I’ve ever figured out a better way to do that than to use a different outbound route when calling my own external number.

I use the same outbound for cell phones because of this problem.

I don’t think that’s what you’re seeing, but it reminds me a little bit of what it sounds like you’re seeing.

It’s possible that you’re internal address is leaking out through your NAT. Make sure all of your NAT settings are correct in the trunk, the outbound route, the extensions, etc.

7

Yes, its the same trunk for inbound and outbound.
The Trunk-settings is:
host=sipconnect.sipgate.de
fromdomain=sipconnect.sipgate.de
outboundproxy=sipconnect.sipgate.de
username=XXXXXXX
fromuser=XXXXXXX
secret=XXXXXXX
port=5060
type=peer
context=from-trunk
canreinvite=no
caninvite=no
dtmfmode=rfc2833
insecure=port,invite
nat=no
registertimeout=600
disallow=all
allow=ulaw
allow=alaw
progressinband=yes

But it makes no diffrence. I tried all possible switches, it changes nothing

8

I don’t know this one. Remove it.
“caninvite=no”

change nat=no to nat=yes

remove disallow=all (just for testing)

9

Already testet, (i’ve tested nearly all possible switches).

10

Hi Dave!

You mean for testing purposes, right (otherwise the easiest solution would be to create a loopback trunk and route…)?

Have a nice day!

Nick

11

I have heard (but not personally verified) that adding 127.0.0.1/24 to local subnets in Asterisk SIP settings can fix this.

12

I guess I might just have to try that. Nothing else I’ve tried in the past 10 years has worked, so that might knock it down.

Actually, I think the “easiest” solution is to do it the way I’ve done it. I have to have the second route anyway (for my cell phone/local prison phone customers) but I’ll look at the loopback route thing and let everyone know.

I’ve tried so many things trying to get this to work that I just don’t remember what didn’t work. It’s been a problem for me since Asterisk 1.4…

13

Hello,

Set a keep alive on the RTP in the sip settings module (rtpkeepalive=1).

image.png816×234 12.4 KB

Thank you,

Daniel Friedman
Trixton LTD.

14

HEY!
This did it. Sowhat strange. I never needed to set this setting so far.

THANKS A LOT!

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